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Noah Richards 3004fd0888 Don't fail SetStereoPlayout(false) for Android devices. 
It isn't implemented, but failing produces warning messages in logs
from code that just does the equivalent of:
SetStereoPlayout(StereoPlayoutIsAvailable)

BUG=none

Specifically:
https://cs.chromium.org/chromium/src/third_party/webrtc/voice_engine/voe_base_impl.cc?l=323

Change-Id: Iad1b026d903bbab74923db35bde50054f125d84b
Reviewed-on: https://chromium-review.googlesource.com/612218
Reviewed-by: Henrik Andreasson <henrika@webrtc.org>
Commit-Queue: Henrik Andreasson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19365}
2017-08-16 11:26:17 +00:00
..
api
Revert of Add a flags field to video timing extension. (patchset #15 id:280001 of https://codereview.webrtc.org/3000753002/ )
2017-08-15 19:31:23 +00:00
audio
Replace absolute path with relative path for GN files.
2017-08-15 15:57:36 +00:00
call
Delete unneeded Start and Stop methods on FlexfecReceiveStream.
2017-08-16 09:41:27 +00:00
common_audio
Replace absolute path with relative path for GN files.
2017-08-15 15:57:36 +00:00
common_video
Revert of Add a flags field to video timing extension. (patchset #15 id:280001 of https://codereview.webrtc.org/3000753002/ )
2017-08-15 19:31:23 +00:00
examples
AppRTCMobile: Use high resolution app icon on retina screens.
2017-08-15 11:07:12 +00:00
logging
Replace CHECK(x && y) with two separate CHECK() calls
2017-08-10 00:22:01 +00:00
media
Delete unneeded Start and Stop methods on FlexfecReceiveStream.
2017-08-16 09:41:27 +00:00
modules
Don't fail SetStereoPlayout(false) for Android devices.
2017-08-16 11:26:17 +00:00
ortc
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
p2p
Replace CHECK(x && y) with two separate CHECK() calls
2017-08-10 00:22:01 +00:00
pc
Revert of Trace the stats report as JSON instead of each stat separately. (patchset #3 id:100001 of https://codereview.webrtc.org/2986453002/ )
2017-08-15 06:48:03 +00:00
rtc_base
Replace absolute path with relative path for GN files.
2017-08-15 15:57:36 +00:00
rtc_tools
Make path to wav file for jitter buffer simulation in event_log_visualizer configurable.
2017-08-16 09:48:33 +00:00
sdk
Revert of Add a flags field to video timing extension. (patchset #15 id:280001 of https://codereview.webrtc.org/3000753002/ )
2017-08-15 19:31:23 +00:00
stats
Audit of kConstants missing the const qualifier
2017-08-08 17:48:15 +00:00
system_wrappers
Replace absolute path with relative path for GN files.
2017-08-15 15:57:36 +00:00
test
Delete unneeded Start and Stop methods on FlexfecReceiveStream.
2017-08-16 09:41:27 +00:00
video
Delete unneeded Start and Stop methods on FlexfecReceiveStream.
2017-08-16 09:41:27 +00:00
voice_engine
Add thread annotations to AudioLevel
2017-08-10 12:01:21 +00:00
.gitignore
…
BUILD.gn
Replace absolute path with relative path for GN files.
2017-08-15 15:57:36 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
common_types.h
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
config_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
config.cc
Audit of kConstants missing the const qualifier
2017-08-08 17:48:15 +00:00
config.h
Audit of kConstants missing the const qualifier
2017-08-08 17:48:15 +00:00
DEPS
Move webrtc/{tools => rtc_tools}
2017-06-30 10:04:59 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
OWNERS
Don't add or rename files in webrtc/ and webrtc/api/ without a proper review
2017-05-08 13:01:54 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
typedefs.h
Make WARN_UNUSED_RESULT a no-op on gcc
2017-04-06 11:32:27 +00:00
video_receive_stream.h
Protected streams report RTP messages directly to the FlexFec streams
2017-08-02 14:39:07 +00:00
video_send_stream.h
Move RTP keep-alive config from VideoSendStream::Config to Call::Config
2017-07-11 10:44:17 +00:00
webrtc.gni
Replace absolute path with relative path for GN files.
2017-08-15 15:57:36 +00:00
whitespace.txt
Moving whitespace file up by one folder
2017-01-24 13:19:42 +00:00

README.chromium 

Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.